Method of preparing data, in particular encoded voice signal par

Details Method of preparing data, in particular encoded voice signal par Method of preparing data, in particular encoded voice signal par

1995-09-25

1998-08-11

MacDonald, Allen R.

Data processing: speech signal processing, linguistics, language

Speech signal processing

For storage or transmission

704229, 704230, G10L 302

Patent

active

057941831

DESCRIPTION:

BRIEF SUMMARY
BACKGROUND OF THE INVENTION

The invention relates to a method of preparing data, in particular encoded voice signal parameters for transmission purposes.
In the encoding and decoding of voice signals, in particular for mobile radio applications, the voice signal is sampled and sub-divided into intervals (time intervals). For each interval, predicted values are formed for different types of signal parameters. Such signal parameters are, for example, short-term parameters for characterizing the formant structure (resonances of the voicebox) and long-term parameters for characterizing the pitch structure (level of tone) of the voice signal (ANT 1988, pages 93-105). In voice encoding by means of "Analysis by Synthesis", the model and excitation parameters are quantized, encoded and transmitted to the receiver. For further reducing the bit rate, vector quantization is used (see above; DE/EP 0 266 620 T1; EP 504 627 A2; EP 294 020 A2).


SUMMARY OF THE INVENTION

The object of the present invention is to develop a method of the type mentioned at the beginning such that, with further reducing of the bit rate, a satisfactory reconstruction of the output data is possible. This object is achieved by the steps of claim 1. The further claims illustrate advantageous refinements.
The method according to the invention is distinguished in particular by its robustness with respect to transmission errors. The method according to the invention makes it possible to construct voice codes of which the voice quality is better than in the case of voice codes with reduction of the quantization stages by multiples of 2. Since transmission errors generally occur several at once, the complexity is reduced along with no deterioration in error correction.
An exemplary embodiment of the invention is now explained in more detail with reference to the drawings.


BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 shows a block diagram of a voice coder which operates by the method of the invention,
FIG. 2 shows the frame structure of two frame intervals for different types of signal parameters.


DESCRIPTION OF PREFERRED EMBODIMENTS

As FIG. 1 shows, voice signals of a voice signal source Q are sampled by means of an A/D converter and analyzed with regard to identical voice signal parameters in an analysis unit A. The analysis unit supplies in each case a set of mutually identical voice signal parameters, for example a set of short-term parameters KP for the formant structure (excitation parameters), a set of long-term parameters LP for the pitch structure and a set of filter weighting parameters FP. With these sets of parameters, predicted values are respectively obtained in predictors PRK, PRL, PRF in a conventional way, for example according to EP 364 647, and are subjected to vector quantization VQ. In a frame-forming unit RA, the quantized signal parameters are combined, to be precise for example such that a frame of a frame period of, for example, 20 msec. comprises 4 frame intervals of a period of in each case 5 msec. In each of these frame intervals there are accommodated identical signal parameters. From at least two of these frame intervals (in the following the handling of in each case two frame intervals is described, but more than two frame intervals can of course also be handled together), bits are then suppressed by means of a bit suppression unit BU. According to the invention, the bit suppression is not carried out individually for each frame interval but for the total number of bits from at least two types of combined identical frame intervals, ie. for example for the total number of bits of the short-term and long-term parameters in a frame of a period of 20 msec. In the bit suppression it is ensured that the quantization stages per frame interval are equally distributed. The number n of the bits to be suppressed is advantageously distributed over the frame intervals in accordance with the relationship m.sqroot.2 g-n, where m indicates the number of identical signal parameters and g indicates the total number of original bits. T

REFERENCES:
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patent: 5233660 (1993-08-01), Chen
patent: 5265167 (1993-11-01), Akamine et al.
ICASSP-88. Copperi, Rule based speech analysis and application of CELP coding; p. 143-146 vol. 1, Apr. 1988.
1991 IEEE International Symposiium on Circuits and Systems. Akamine et al., "Efficient Excitation model fro low bit rate speech coding", p. 586-589 vol. 1, Jun. 1991.

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